Microsoft Lync 2013 Skype for Business 2015 · Microsoft Lync 2013 Skype for Business 2015 ... From...

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Microsoft Lync 2013 Skype for Business

2015 Configuration Checklists

for BTIP and Business Talk SIP services

26 april 2017

Lync 2013 Checklist version 1.6

Skype for Business 2015 Checklist version 1.10

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Contents

1 Lync 2013 Configuration Checklist ....................................................................................... 3

2 Skype for Business 2015 Configuration Checklist .............................................................. 17

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1 Lync 2013 Configuration Checklist

Menu Value

DNS requirements

From the DNS interface:

Start > Administrative Tools > DNS

FQDNs of each server (DNS A record)



FQDNs of both nominal and backup aSBC on each site (DNS A record)



ucupdates-r2.<SIP domain> (DNS A record) that maps the FQDN of each

server hosting Device Update Service



_sipinternaltls._tcp.<SIP domain> (DNS SRV record/Port 5061) that maps

the FQDN of each server offering automatic client sign-in service



_ntp._udp.<SIP domain> (DNS SRV record/Port 123) that maps the FQDN

of the Domain Controller

DHCP requirements

From the customer interface of the router Following command has to be typed for each customer interface of the

router:

ip helper-address “IP@ of the DHCP Server”

From the Microsoft Lync Server Management

Shell interface:

Start > All Programs > Microsoft Lync

Server 2013 > Lync Server

Management Shell

Following command has to be typed:

Set-CsRegistrarConfiguration –EnableDHCPServer $True

From the DHCP interface:

Start > Administrative Tools > DHCP

> “select a scope” > Scope Options

DHCP Option 006 DNS Servers has to be activated




“DHCPUtil.exe” and “DHCPConfigScript.bat” files* have to be added

on a network share that can be accessed from the DHCP server

(*) DHCP Options 120 / 43 have to be configured (only if required by the

type of endpoints deployed)

From command prompt from the DHCP

server:

Start > Run… > cmd

Following command has to be typed*:

\\<FileShare>\DHCPUtil.exe -SipServer “SipServer” -WebServer “WebServer” –RunConfigScript

(*) DHCP Options 120 / 43 have to be configured (only if required by the type of endpoints deployed)




DHCP Option 042 NTP Servers has to be activated*

(*) only if required by the type of endpoints deployed

AD requirements

From the AD interface:

Start > Administrative Tools > Active

Directory Users and Computers

Each server role has to be joined to domain

Mediation Server Configuration

From the Microsoft Lync Server Topology

Builder interface:



Topology Builder

Lync Server 2013 > “select a Central

TCP listening port has to be set to 5060

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Menu Value

Site” > Mediation pools > “select a

Mediation Server”

Enterprise Edition – Standalone Mediation Servers - Configuration

From the standalone Mediation Server:

Start > Control Panel > Network and

Internet > Network Connections >

“select the interface of the

Mediation Server” > Properties >

Internet Protocol Version 4

(TCP/IPv4)

Default gateway has to be filled

Preferred DNS server has to be filled

From the standalone Mediation Server:

Start > Control Panel > Network and

Internet > Network Connections >

“select the interface of the

Mediation Server” > Properties >

Internet Protocol Version 4

(TCP/IPv4) > Advanced… > DNS

tab

Register this connection’s addresses in DNS has to be checked


Builder interface:



Topology Builder

Lync Server 2013 > “select an

Enterprise Edition Central Site” >

Mediation pools

2 Mediation pools have to be created for 2 Standalone Mediation Servers:

Multiple computer pool with the Standalone Mediation Server pool 1

(=FQDN of the Mediation Server pool 1)

Multiple computer pool with the Standalone Mediation Server pool 2

(=FQDN of the Mediation Server pool 2)

Enable TCP port has to be checked

Listening port has to be set to 5060 for each standalone Mediation Server

pool


Builder interface:



Topology Builder



Shared Components > PSTN

gateways

2 PSTN gateways have to be created

1st: FQDN of Nominal aSBC (Mediation server pool 1)

2nd: FQDN of Backup aSBC (Mediation server pool 1)

Check that Use all configured IP addresses is selected for each Mediation

Server:

Enable IPv4 has to checked and Enable IPv6 has to be unchecked for each

Mediation Server

Next window contains the Trunk root information as followed

Listening port for IP/PSTGN gateway has to be set to 5060

SIP Transport Protocol has to be set to TCP

Associated Mediation Server has to match the FQDN of Mediation Server

pool 1


Builder interface:



Topology Builder



Shared Components > Trunks

2 Additional Trunks have to be created

1st: Associated PSTN gateway of Nominal aSBC (Mediation server

pool 2)

2nd: Associated PSTN gateway of Backup aSBC (Mediation server

pool 2)

Listening port for IP/PSTGN gateway has to be set to 5060

SIP Transport Protocol has to be set to TCP

Associated Mediation Server has to match the FQDN of Mediation Server

pool 2

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Menu Value

From the Microsoft Lync Server Control Panel

interface:


Server 2013 > Lync Server Control

Panel

Voice Routing > Route

4 Routes have to be created for 2 Standalone Mediation Servers*:

from Standalone Mediation Server 1a to nominal aSBC (=FQDN of

the nominal aSBC from the Mediation Server 1a)

from Standalone Mediation Server 1b to backup aSBC (=FQDN of

the backup aSBC from the Mediation Server 1b)

from Standalone Mediation Server 2a to nominal aSBC (=FQDN of

the nominal aSBC from the Mediation Server 2a)

from Standalone Mediation Server 2b to backup aSBC (=FQDN of

the backup aSBC from the Mediation Server 2b)

A gateway (=FQDN of the nominal aSBC from the Mediation Server 1a) has

to be associated to First Route

A gateway (=FQDN of the backup aSBC from the Mediation Server 1b) has

to be associated to Second Route

A gateway (=FQDN of the nominal aSBC from the Mediation Server 2a) has

to be associated to Third Route

A gateway (=FQDN of the backup aSBC from the Mediation Server 2b) has

to be associated to Fourth Route

A PSTN Usage has to be associated to each Route

(*) Routes for a site Headquarter includes its Remote Sites without MGW

Enterprise Edition – Standalone Mediation Servers – Specific configuration for Remote Site deployment


Builder interface:



Topology Builder

Lync Server 2013 > “select a Branch

Sites” > Lync Server 2013 >

Shared Components > PSTN

gateways

2 PSTN gateways have to be created for the Standalone Mediation Server:

to nominal aSBC (=FQDN of the nominal aSBC)

to backup aSBC (=FQDN of the backup aSBC)

Check that 2 Trunks were created while creating PSTN gateways

Listening port has to be set to 5060 for each PSTN gateways

SIP transport protocol has to be set to TCP for each PSTN gateways


Builder interface:



Topology Builder

Lync Server 2013 > “select a Branch

Sites” > Mediation pools

A Mediation pools has to be configured for the Standalone Mediation Server:

One single computer pool (=FQDN of the Mediation Server)

2 PSTN Gateways have to be associated to the Standalone Mediation

Server:

FQDN of the nominal aSBC

FQDN of the backup aSBC

Use all configured IPv4 IP addresses has to be checked:

Listening port has to be set to 5060


interface:



Panel

Voice Routing > Dial Plan

A Site dial plan has to be created for each Remote site with a Standalone

Mediation Server

A New Normalization Rule for extension numbers has to be associated:

Pattern to match has to be edited

Translation rule has to be edited

Internal extension has to be checked

Normalization Rule for extension numbers has to be moved up before the

existent Normalization Rule for Prefix All


interface:



Panel

Voice Routing > Voice Policy

An User policy has to be created for each Remote site with a Standalone

Mediation Server

Enable call park has to be checked

Enable PSTN reroute has to be unchecked

A PSTN Usage has to be associated to each User policy

From the Microsoft Lync Server Control Panel The specific voice policy has to be assigned to each RS (with a Standalone

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Menu Value

interface:



Panel

Users > “select an user of Remote

Site with a Standalone Mediation

Server”

Mediation Server) user


interface:



Panel


2 Routes have to be created for each Remote site with a Standalone

Mediation Server :

to nominal aSBC

to backup aSBC

A gateway (=FQDN of nominal aSBC) has to be associated to First Route

A gateway (=FQDN of backup aSBC) has to be associated to Second Route



interface:



Panel

Voice Routing > Trunk Configuration

A Site trunk has to be created for each Remote site with a Standalone

Mediation Server

Enable refer support has to be unchecked

Encryption support level has to be set to Optional

A Translation Rule (to remove digit “+” for outbound calls to BTIP SIP) has to

be associated to each Site trunk


Shell interface:



Management Shell

Following commands have to be typed for each Remote site with a

Standalone Mediation Server:

Set-CsTrunkConfiguration –Identity “Site” –RTCPActiveCalls

$False

Set-CsTrunkConfiguration –Identity “Site” –RTCPCallsOnHold

$False


interface:



Panel


A PSTN Usage of Branch Sites has to be associated to each Route of

Headquarter

Note that routes must be in the following order:

1) Route of Branch Sites to nominal aSBC

2) Route of Branch Sites to backup aSBC

3) Route of Headquarter to nominal aSBC

4) Route of Headquarter to backup aSBC

Users Configuration

From the AD interface:

Start > Administrative Tools > Active

Directory Users and Computers

New > User

User information (the user logon name) has to be filled


interface:



Panel

Users > Enable users > Add… > Find

Each user has to be assigned to a pool

Format <SAMAccountName>@<SIP domain> has to be selected

Telephony has to be set to Enterprise Voice

An E164 telephone number format followed by an extension number has to

be entered in the line URI

Routing mechanisms for Microsoft Lync Server 2013


interface:



Panel


A Site dial plan has to be created for each site

A New Normalization Rule for extension numbers has to be associated:

Pattern to match has to be edited

Translation rule has to be edited

Internal extension has to be checked

Normalization Rule for extension numbers has to be moved up before the

existent Normalization Rule for Prefix All

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Menu Value

(*) Site dial plan for a site Headquarter includes its Remote Sites without MGW


interface:



Panel


A Site policy has to be created for each site*



A PSTN Usage has to be associated to each Site policy

(*) Site policy for a site Headquarter includes its Remote Sites without MGW


interface:



Panel


2 Routes have to be created for each site* :

to nominal aSBC

to backup aSBC

A gateway (=FQDN of nominal aSBC) has to be associated to First Route

A gateway (=FQDN of backup aSBC) has to be associated to Second Route


(*) Routes for a site Headquarter includes its Remote Sites without MGW


interface:



Panel


A Site trunk has to be created for each site*

Enable refer support has to be unchecked

Enable forward call history has to be checked

Encryption support level has to be set to Optional

A Translation Rule (to remove digit “+” for outbound calls to BTIP SIP) has to

be associated to each Site trunk

(*) Site trunk for a site Headquarter includes its Remote Sites without MGW


Shell interface:



Management Shell

Following commands have to be typed for each site*:

Set-CsTrunkConfiguration –Identity “Site” –RTCPActiveCalls

$False

Set-CsTrunkConfiguration –Identity “Site” –RTCPCallsOnHold

$False

(*) A Site Headquarter includes its Remote Sites without MGW


Shell interface:



Management Shell

Following command has to be typed:

Set-CsMediaConfiguration –EncryptionLevel

SupportEncryption

Specific Normalization Rule

Voice Mail Feature :


interface:

Start > All Programs > Microsoft Lync Server

2013

> Lync Server Control Panel


A Normalization Rule has to be associated to each Site dial plan*

(*) to be adapted according the client architecture

Call Park Feature :


interface:


2013

> Lync Server Control Panel


A Normalization Rule has to be associated to each Site dial plan*

(*) to be adapted according the client architecture

Music On Hold

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Menu Value


Shell interface:



Management Shell

Note:

The customized MoH is played For

Softphone Devices

The embedded firmware MoH is played For

Lync Phone Edition Devices

The global clientpolicy is used:

Following commands have to be typed for Softphones

New-CsClientPolicy –Identity global –EnableClientOnHold

$True –MusicOnHoldAudioFile <FILE PATH> Note:

No more need to associate Each user to a specific Client Policy, check only

while user creation that client policy field is set to Automatic

Unified Messaging on Microsoft Exchange Server 2013

From the Exchange Server Administration Url:

https://exchangeserverIPaddress/ecp

logon using administrator credential

Select Unified Messaging

Double click on UM DialPlan then click

on configure

On the General tab, VoIP security has to be set to Secured


Shell interface:



Management Shell

Following command has to be typed

Set-UMservice –Identity <ExchangeServer> –UMStartUpMode TLS






on configure

On the Settings tab, Audio codec has to be set to GSM






on configure

On the Outlook Voice Access, A Subscriber Access Number (E164

telephone number format) has to be added

From the Exchange UM server (Config file):

C:\Program Files\Microsoft\Exchange

Server\V15\Bin\MSExchangeUM

<add key="MinimumRtpPort" value="49152" />

<add key="MaximumRtpPort" value="57500" />

From the Exchange UM server (Local Group

Policy Editor ):

Start > Run… > gpedit.msc

Audio Policy-based QoS is configured

Source port: 49152:57500

Protocol: TCP and UDP

DSCP: 46

From the Front End Server:

C:\Program Files\Common

Files\Microosft Lync Server

2013\Support\OcsUmUtil.exe

On the OcsUmUtil tool:

Click Load Data

Double click on contacts

Select Use this pilot number from Exchange UM has to match the

subscriber access number (E.164 telephone number format)

Analog Devices Configuration

From the Microsoft Server 2013 Control Panel and Management Shell


interface:

An User policy has to be created for each site with Analog Devices

https://exchangeserveripaddress/ecp



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Menu Value



Panel




An Existent PSTN Usage has to be associated by selecting it


Shell interface:



Management Shell

Following command has to be typed for each Analog Device :

New-CsAnalogDevice “LineURI” –DisplayName

“DisplayName” –RegistrarPool “RegistrarPool” –AnalogFax

$False –Gateway “Gateway” –OU “OU”


Shell interface:



Management Shell

Following command has to be typed for each Analog Device :

Set-CsAnalogDevice -Identity “Identity” –DisplayNumber

“DisplayNumber” Set-CsAnalogDevice -Identity “Identity” –LineURI “LineURI” Grant-CsVoicePolicy -Identity “Identity” –PolicyName

“PolicyName”

From the Sonus (NET) (UX 1000/2000 SBA)

From the UX Web User interface:

Settings Tab > Media > Media List

A Media List has to be created:

Media List for Analog Devices:

Media Profiles List has to match the Voice Codec Profile G711 A-Law

Digit Relay

Digit (DTMF) Relay Type has to be set to RFC 2833

Digit Relay Payload Type has to be set to 101


Settings Tab > CAS > CAS Signaling

Profiles

A FXS CAS Signaling Profiles has to be created


Settings Tab > Signaling Groups

A CAS Signaling Group has to be created:

CAS Signaling Group for Analog Devices connectivity:

CAS Protocol

CAS Signaling Profile has to match the CAS Signaling Profile for Analog

Devices

Channels and Routing

Channel Hunting has to be set to Own Number

Tone Table has to match the Analog Device Tone Table

Call Routing Table has to match the Analog Device Call Routing Table**

for routing calls received from Analog Devices

Assigned Channels

Channel Phone Number has to match the Analog Device phone number

(**) Please note that Call Routing Table must be added later (after specific

Call Routing Tables configuration)


Settings Tab > Transformation

A Transformation Table has to be created:

Transformation Table for Lync to Analog Device calls:

Input Field

Value has to match the Analog Device telephone number E.164 format

Output Field

Value has to be set to \1


Settings Tab > Call Routing Table

A Call Routing Table has to be created for calls received from Lync (if it

doesn’t exist) or additionals Call Routing Entries have to be created in the

Call Routing Table for calls received from Lync (if it exists)

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Menu Value

Call Routing Entry for Lync to Analog Device calls:

Route Details

Number/Name Transformation Table has to match the Transformation

Table for Lync to Analog Device calls

Destination Information

Destination Signaling Groups has to match the Signaling Group for

Analog Device connectivity

Media

Media List has to match the Media List for Analog Device

A Call Routing Table has to be created for calls received from the Analog

Devices

Call Routing Entry Tenor to Lync calls:

Route Details


Table used to send a phone number without modification


Destination Signaling Groups has to match the Signaling Group for Lync

connectivity

Media

Media List has to match the Media List for Analog Device

(**) Please note that Call Routing Table must be added to CAS Signaling

Groups configuration

From the AudioCodes (Mediant 800/1000 SBA)

From the AudioCodes Web User interface:

Configuration Tab (full) >VoIP menu >

TDM submenu > Select TDM Bus

Settings

PCM Law Select has to be set to A-Law

TDM Bus Clock Source has to be set to Network



Media submenu > Select Voice

Settings



Media submenu > Select Analog

Settings

CAS Transport Type has to be set to CASRFC2833Relay

Check that Analog Settings are filled with default value



Coders and Profiles submenu >

Select Analog Coders

Coder Name has to be set to G711 A-Law

Packetization Time has to be set to20ms

Payload Type has to be set to 8



GW and IP to IP submenu > Trunk

Group > Select Trunk Group

A Trunk Group has to be created with the following parameters:

Module has to be set to Module 2 FXS

Channels has to be set to the Analog Device port on the gateway

Phone Number has to match the Analog Device

phone number

Trunk Group ID has to match the Analog Device

Trunk Group ID

Tel Profile ID has to match the Tel Profile ID if configured else the default

profile 0 has to be associated

Trunk Group ID has to match the Analog Device

Trunk Group ID

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Menu Value





Settings

Channel Select Mode has to be set to By Dest Phone Number



GW and IP to IP submenu >

Manipulation > Select Dest Number

IP -> Tel





Tel -> IP

Destination Prefix has to match the Analog Device

Phone Number as declared on the Trunk Group Table

Source Trunk Group has to match the Analog Device

Trunk Group already created

Prefix to add has to match a rule manipulation in order to has a E.164

format number to send to Lync Server




Routing > Select Tel to IP Routing




Routing > Select IP to Tel Routing

Tel to IP Routing Mode has to be set to Route Calls after manipulation

Src IP Group ID has to be set to -1

Src Trunk Group ID has to match the Analog Device Group ID

Dest IP Group ID has to match the Lync Server Group ID

IP toTel Routing Mode has to be set to Route Calls before manipulation

Dest Phone Prefix has to match the Analog Device phone number

Trunk Group ID has to match the Analog Device Trunk Group ID

IP Profile ID has to match the Tel Profile ID if configured else the default


E1/T1 Access Configuration

From the Sonus (NET) (UX 1000/2000 SBA) with FXS ports


Settings Tab > Signaling Groups

An ISDN Signaling Group has to be created:

ISDN Signaling Group for E1/T1 connectivity:

Port and Protocol

Port Name has to be selected

Switch Variant has to be set to Euro ISDN

Channels and Routing

Tone Table has to match the Tone Table if configured else the Default

Tone Table has to be selected

Call Routing Table has to match the E1/T1 Call Routing Table** for

routing calls received from E1/T1 access

(**) Please note that Call Routing Table must be added later (after specific

Call Routing Tables configuration)



Transformation Table for T2 to Lync calls


Transformation Entry for T2 to Lync calls (Called):

Input Field

Type has to be set to Called Address/Number

Value has to match the T2 number

Output Field


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Menu Value



Value has to match the E.164 Lync number

Transformation Entry for T2 to Lync calls (Calling):

Input Field

Type has to be set to Calling Address/Number

Value has to be filled

Output Field



Transformation Table for Lync to T2 calls


Transformation Entry for Lync to T2 calls (Called):

Input Field



Output Field



Transformation Entry for Lync to T2 calls (Calling):

Input Field



Output Field




Settings Tab > Call Routing Table

Call Routing Table for Lync to T2 calls

A Call Routing Table has to be created for calls received from Lync (if it

doesn’t exist) or an additional Call Routing Entry has to be created in the Call

Routing Table for calls received from Lync (if it exists)

Call Routing Entry for Lync to T2 calls:

Route Details


Table for Lync to T2 calls


Destination Signaling Groups has to match the Signaling Group for

E1/T1 connectivity

Media

Media List has to match the Media List without crypto

Call Routing Table for T2 to Lync calls

A Call Routing Table has to be created for calls received from E1/T1 access

Call Routing Entry for T2 to Lync calls:

Route Details


Table T2 to Lync calls

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Menu Value


Destination Signaling Groups has to match the Signaling Group for Lync

connectivity

Media

Media List has to match the Media List without crypto

(**) Please note that Call Routing Table must be added to ISDN/SIP

Signaling Groups configuration

From AudioCodes Mediant (800/ 1000 SBA)



PSTN submenu > Select Trunk

Settings

Protocol Type has to be set to E1 Euro ISDN

Line Code has to be set toHDB3

Framing Method has to be set to E1 FRAMING MFF CRC4 EXT









Settings

A Trunk Group has to be created with the following parameters:

Module has to be set to Module 1 PRI

Channels has to be set to T2 line number of channels

Phone Number has to match the T2

phone number

Trunk Group ID has to match the T2

Trunk Group ID

Tel Profile ID has to match the Tel Profile ID if configured else the default


Trunk Group ID has to match the T2

Trunk Group ID

Channel Select Mode has to be set to Cyclic Ascending



Control Network submenu > Select

Proxy Set Table

A Proxy Set Table has to be created with the following parameters:

Proxy Set ID has to be filled

Proxy Address has to match the SBA FQDN

Transport Type has to be set to TLS

Enable Proxy Keep Alive has to be set to Using Options



Control Network submenu > Select

IP Group Table

An IP Group Table has to be created with the following parameters:

Index has to be filled

Type has to be set to Server

Proxy Set ID has to match the SBA proxy Set ID already created





IP -> Tel





Tel -> IP

Destination Prefix has to be filled with the prefix of the received number

Source IP Address has to match the SBA IP Address

Stripped Digits from Left has to be filled

Prefix to Add has to be filled

Source Trunk Group has to match the T2 Trunk Group already created

Destination Prefix has to match the T2 Line number

Stripped Digits from Left has to be filled

Prefix to add has to match the corresponding Lync device on E.164

format number




Tel to IP Routing Mode has to be set to Route Calls after manipulation

Src IP Group ID has to be set to -1

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Menu Value

Routing > Select Tel to IP Routing




Routing > Select IP to Tel Routing

Src Trunk Group ID has to match the T2 Group ID

IP toTel Routing Mode has to be set to Route Calls before manipulation

Source IP Address has to match the Gateway IP Address

Trunk Group ID has to match the T2 Trunk Group ID

IP Profile ID has to match the Tel Profile ID if configured else the default


Dial-in Conferencing feature


interface:



Panel


A Dial-in conferencing region has to be added (associated to Dial-in Access

Number)

Call Back feature


interface:



Panel


A specific translation Rule has to be associated to each Site trunk

(*) to be adapted to the client architecture

(**) first priority before translation rule removing the « + » digit

Call Park feature


interface:



Panel

Voice Features

A Number range has to be created for each Site

(*) to be adapted to the client architecture

CALL ADMISSION CONTROL


interface:



Panel

Network Configuration > Global

Edit Global Setting –Global

Check Enable call admission control


interface:



Panel

Network Configuration > Bandwidth Policy

Create Bandwidth Policy for CAC “from site to WAN”

New “name”

Audio limit: according to site sizing

Audio session limit: 100

Create Bandwidth Policy for CAC “from Edge to WAN”

New “name”



Create Bandwidth Policy for CAC “from site to SIP Trunk”

New “name”



Create Bandwidth Policy for CAC “0”

New “name”

Audio limit: 0


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Menu Value


interface:



Panel

Network Configuration > Region

Create WAN Region

New “name”

Associate site name

Uncheck Enable audio alternate path (recommended)

Check or Uncheck Enable video alternate path to your convenience


interface:



Panel

Network Configuration > Site

Create Site for users and associate a Bandwidth policy between this Site

and the Region

New “name”

Associate Region

Associate Bandwidth Policy for CAC “from site to WAN”

Create Site for edge and associate a Bandwidth policy between this Site and

the Region

New “name”

Associate Region

Associate Bandwidth Policy for CAC “from Edge to WAN”

Create Site for aSBC and associate a Bandwidth policy between this Site

and the Region

New “name”

Associate Region

Associate Bandwidth Policy for CAC “0”


Shell interface:


2013 > Lync Server Management Shell

Creation of Bandwidth Policy for intersite links

New-CsNetworkInterSitePolicy –Identity “name of the intersitelink” -BWPolicyProfileID “name of the policy for CAC from site to SIP Trunk” -NetworkSiteID1 “name of the site for user” -NetworkSiteID2 “name of the sitefor the SBC”


interface:



Panel

Network Configuration > Subnet

Create subnet for each site

New

Add subnet ID

Add mask

Associate with Network site ID

Quality of Service

From the Microsoft Lync Management Shell

interface::



Management Shell

Enable client media port range:

Set-CsConferencingConfiguration –ClientMediaPortRangeEnabled $true

–ClientMediaPort 50000 –ClientAudioPort 50060 –ClientVideoPort

57600 –ClientAppSharingPort 32800


interface::



Management Shell

Configure ApplicationSharing port range on Lync application servers:

Set-CsApplicationServer ApplicationServer:<serverFQDN> -

AppSharingPortStart 32768 –AppSharingPortCount 16383


interface::



Management Shell

Configure ApplicationSharing port range on Lync Conferencing servers:

Set-CsApplicationServer ConferencingServer:<serverFQDN> -

AppSharingPortStart 32768 –AppSharingPortCount 16383

Configuration requirements (warnings)

Configuring Clients ports range for LPE and SoftPhone

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Menu Value


interface::



Management Shell

Enable client media port range:

Set-CsConferencingConfiguration –ClientMediaPortRangeEnabled $true

–ClientAudioPort 50060 –ClientAudioPortRange 48

Configuring Clients ports range for VVX

Using VVX Web UI Navigate through the VVX Web Interface: http:<VVX_IP_Address>

Go to Settings tab > Network menu > RTP

Configure the Port Range Start to: 50060

Using VVX configuration file (.cfg) Configure the following line in the VVX configuration file :

tcpIpApp.port.rtp.mediaPortRangeStart="50060"

Import the new configuration file to the VVX using the WebUI or through the

IIS server

Others Devices

Check that the audio range port respect the

OBS recommendations

The default audio range is: 50060-50107.

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2 Skype for Business 2015 Configuration Checklist

Menu Value

Skype for Business Configuration (Topology Builder)

On the Topology builder interface:

Central Site > skype for business 2015 > Mediation Pools, right click and Edit properties

Enable TCP port has to be checked

Listening port has to be set to 5060 for

each Mediation Server in skype for Business topology


Central Site > Skype for Business 2015 > Shared components > Trunks, right click edit properties

FQDN of nominal aSBC for BT/BTIP traffic

Specify nominal aSBC BT/BTIP trunk name

Listening port for IP/PSTN gateway: 5060

SIP Transport protocol: TCP

Associated Mediation Server: Mediation Server FQDN

Associated Mediation Server port: 5060


Central Site > Skype for Business 2015 > Shared components > Trunks, right click edit properties

FQDN of backup aSBC for BT/BTIP traffic

Specify backup aSBC BT/BTIP trunk name



Associated Mediation Server: Mediation Server FQDN


Skype for Business Configuration (Control Panel)

Dial Plan

On the Skype for Business Server Control Panel Interface:


Type: Dial Plan type

Name: Dial Plan name

Voice Policy



Name: Voice Policy name

Enable call park: Checked

Enable PSTN reroute: Unchecked

PSTN usage



New PSTN Usage record

Name: BT/BTIP PSTN Usage name

Routes (aSBC nominal route)



Edit PSTN Usage record

Associated routes New

Name: aSBC nominal Route name

Associated Trunks Add

Select corresponding aSBC nominal Trunk

from drop down list

Routes (aSBC backup route)





Name: aSBC backup Route name


Select corresponding aSBC backup Trunk from drop down list

Trunk configuration


Voice Routing > Trunk configuration

New

Name: BT/BTIP Trunk name

Encryption support level : Optional

Refer support : None

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Menu Value

Enable forward call History : Checked

Trunk configuration (SFB PowerShell)

On the Skype for Business PowerShell Interface:

Set-CsTrunkConfiguration –Identity <Site> –RTCPActiveCalls $False

Set-CsTrunkConfiguration –Identity <Site> –RTCPCallsOnHold $False

-Site: The name of the site

Configuration Checklist in case of Sonus SBC 1000/2000 Gateway:

This configuration checklist will follow this color convention:

Green: in case of RS SBA

Blue: in case of HQ with GW aboard

Skype for Business– RS SBA or HQ with GW aboard - Trunk SIP on sonus SBC BT/BTIP configuration

PSTN usage

On the Skype for Server Control Panel Interface:


New sonus SBC BT/BTIP PSTN Usage record

Name: sonus SBC BT/BTIP PSTN Usage name

Route (sonus SBC BT/BTIP)





Name: sonus SBC for BT/BTIP route name


Select corresponding sonus SBC Trunk from drop down list

Trunk configuration



New

Name: sonus SBC for BT/BTIP Trunk name








-Site: The name of the remote site

Sonus SBC BT/BTIP configuration

SIP Profile

On the Sonus SBC gateway WebUi Interface:

Settings >SIP > SIP Profile > Default SIP Profile

Session Timer:

Session Timer: Disabled

Header Customization:

UA Header: Sonus SBC

Calling Info Source: RFC Standard

Options Tags:

100rel: Supported

Update: Supported

SDP Customization:

Send Number of Channels: True

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Menu Value

Connection Info In Media Section: True

Digit Transmission Preference: RFC 2833/Voice

Media


Settings >Media > Media System Configuration

Port Range:

Start Port: 16384

Number of Port pairs: 600

Echo Canceller Type Option: Standard

Echo Cancel NLP Option: Mild

Send STUN Packets: Enabled

Music On Hold:

Music on Hold Source: File


Settings >Media > Media Profiles

Default G711a:

Codec: G711 A-law

Payload Size: 20 ms

Default G711µ:

Codec: G711 µ-law

Payload Size: 20 ms


Settings >Media > Media List

Default Media List:

Media Profiles List: G711a

G711µ

Crypto Profile ID: None

Media DSCP: 46

RTCP Mode: RTCP

Dead Call Detection: Disabled

Silence Suppression: Disabled

Secondary interface (only for RS SBA)


Settings >Node Interfaces > Logical Interfaces > Ethernet 1 IP

Configure Secondary Interface: Enabled

Secondary Address: IP address of the secondary interface of the Sonus gateway (dedicated for BT/BTIP traffic)

Secondary Mask: Mask corresponding to secondary interface subnet

From/To SFB <-> Offnet routing BT/BTIP traffic

SIP Server Table

From/To SBA –BT/BTIP or From/To MS Pool –BT/BTIP


Settings >SIP > SIP Server Tables > Create SIP Server

Host: SBA or MS Pool IP address

Port: 5060

Protocol: TCP

Monitor: SIP Options

From/To BT/BTIP-SBA or From/To MS Pool –BT/BTIP



1st Entry: ACME aSBC nominal

Host: ACME aSBC nominal IP address

Port: 5060

Protocol: TCP


2nd Entry: ACME aSBC backup

Host: ACME aSBC backup IP address

Port: 5060 Protocol: TCP


Transformation Rules

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Menu Value

SBA to BT/BTIP or MS Pool to BT/BTIP


Settings >Transformation > New Transformation Table > New Transformation Entry

Calling Entry:

Input Field Type: Calling Address/Number

Input Field Value: depend on transformation

need

Output Field Type: Calling Address/Number

Output Field Value: depend on transformation

need

Called Entry:

Input Field Type: Called Address/Number


need

Output Field Type: Called Address/Number


need

BT/BTIP to SBA or BT/BTIP to SBA



Calling Entry:



need



need

Called Entry:


Input Field Value: must normalize received

number on Skype for Business E.164 number

format


Output Field Value: depend on transformation need

Call Routing Tables

From SBA or From MS Pool


Settings >Call Routing Table > Create

SBA to BT/TIP or MS Pool to BT/TIP entry:

Description: SBA to BT/BTIP or MS pool to BT/BTIP

Route Priority: 1

Number/Name Transformation Table: SBA to

BT/BTIP or MS Pool to BT/BTIP

Destination Signalling Group: (SIP) From/To

BT/TIP-SBA or From/To BT/TIP-SBA

Media Transcoding: Enabled (If licenced)

From BT/BTIP



BT/TIP to SBA or BT/TIP to MS Pool entry:

Description: BT/BTIP to SBA or BT/BTIP to MS Pool

Route Priority: 1

Number/Name Transformation Table:

BT/BTIP to SBA or BT/BTIP to MS Pool

Destination Signalling Group: (SIP) From/To SBA-BT/BTIP or From/To MS Pool-BT/BTIP


Signaling Groups

(SIP) From/To SBA – BT/BTIP or From/To MS Pool – BT/BTIP


Settings >Signaling Group > SIP Signaling Group

Description: SIP From/To SBA – BT/BTIP or From/To MS Pool – BT/BTIP

Call Routing Table: From SBA or From MS Pool

SIP Server Table: From/To SBA –BT/BTIP or

MS Pool –BT/BTIP

Signalling/Media Source IP :Sonus BT/BTIP

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Menu Value

interface IP address

Listen Ports:5060 /TCP

Federated IP/FQDN: SBA or MS Pool FQDN

(SIP) From/To BT/BTIP-SBA or From/To BT/BTIP-MS Pool



Description: SIP From/To BT/BTIP-SBA or From/To BT/BTIP-MS Pool

Call Routing Table: From BT/BTIP

SIP Server Table: From/To BT/BTIP -SBA or

From/To BT/BTIP-MS Pool

Signalling/Media Source IP: Sonus BT/BTIP



Federated IP/FQDN: ACME aSBC nominal IP

address

ACME aSBC backup IP

address

From/To SFB <-> Offnet routing E1/T1 traffic (only for RS SBA)

System Companding Law


Settings >System > System companding law

Companding law: A-Law

SIP Server Table

From/To SBA –PSTN



Host: SBA IP

Port: example 5060 (must be the same as defined on Skype for Business topology builder)

Protocol: TCP


Note:

If using same protocol and port as BT/BTIP

the same SIP Server table can be used


SBA to PSTN



Calling Entry:



need



need

Called Entry:



need



need

PSTN to SBA



Calling Entry:



need



need

Called Entry:




format

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Menu Value



Call Routing Tables

From SBA



SBA to PSTN entry:

Description: SBA to PSTN

Route Priority: 1


PSTN

Destination Signalling Group: (ISDN) From/To

PSTN-SBA


From PSTN



PSTN to SBA entry:

Description: PSTN to SBA

Route Priority: 1

Number/Name Transformation Table: PSTN

to SBA

Destination Signalling Group: (SIP) From/To SBA-PSTN


Signaling Groups

(SIP) From/To SBA – PSTN



Description: SIP From/To SBA – PSTN

Call Routing Table: From SBA

SIP Server Table: From/To SBA –PSTN

Signalling/Media Source IP :Sonus E1/analog



Federated IP/FQDN: SBA IP address

(ISDN) PSTN


Settings >Signaling Group > Signaling Group > ISDN Signaling Group

Description: ISDN PSTN

Switch variant: Euro ISDN

Call Routing Table: From PSTN

From/To SFB <-> Offnet routing Analog Devices traffic

SIP Server Table

From/To SBA –Analog Device



Host: SBA FQDN/IP address

Port: example 5060 (must be the same as defined on Skype for Business topology builder)

Protocol: TCP


If using same protocol and port as BT/BTIP

the same SIP Server table can be used ( no

need to create a new SIP Server table)


SBA to Analog



Calling Entry:



need


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Menu Value


need

Called Entry:



need



need

Analog Device to SBA



Calling Entry:



need



need

Called Entry:




format



Call Routing Tables

From SBA



SBA to analog device entry:

Description: SBA to Analog Device

Route Priority: 1


PSTN

Destination Signalling Group: (CAS) Analog

Device


From Analog Device



Analog Device to SBA entry:

Description: Analog Device to SBA

Route Priority: 1

Number/Name Transformation Table: Analog

Device to SBA

Destination Signalling Group: (SIP) From/To SBA-Analog Device


Signaling Groups

(SIP) From/To SBA – Analog Device



Description: SIP From/To SBA – Analog Device

Call Routing Table: From SBA

SIP Server Table: From/To SBA –Analog

Device

Signalling/Media Source IP :Sonus E1/analog



Federated IP/FQDN: SBA IP address

(CAS) Analog



Description: CAS Analog

CAS Signalling Profile: CAS Analog

Call Routing Table: Analog to SBA

Assigned Channels: Analog Devices information

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Menu Value

Skype for Business– RS GW BT/BTIP configuration

PSTN usage

On the Skype for Server Control Panel Interface:


New sonus SBC BT/BTIP PSTN Usage record

Name: sonus Gateway BT/BTIP PSTN Usage name

Route (sonus SBC BT/BTIP)





Name: BT/BTIP Sonus GW route name


Select corresponding sonus GW Trunk from drop down list

Trunk configuration



New

Name: sonus SBC for BT/BTIP Trunk name




Enable media bypass : Checked






Sonus GW BT/BTIP configuration

SIP Profile


Settings >SIP > SIP Profile > Default SIP Profile

Session Timer:

Session Timer: Disabled

Header Customization:

UA Header: Sonus SBC

Calling Info Source: RFC Standard

Options Tags:

100rel: Supported

Update: Supported

SDP Customization:

Send Number of Channels: True

Connection Info In Media Section: True

Digit Transmission Preference: RFC 2833/Voice

Media


Settings >Media > Media System Configuration

Port Range:

Start Port: 16384

Number of Port pairs: 600

Echo Canceller Type Option: Standard

Echo Cancel NLP Option: Mild

Send STUN Packets: Enabled

Music On Hold:

Music on Hold Source: File


Settings >Media > Media Profiles

Default G711a:

Codec: G711 A-law

Payload Size: 20 ms

Default G711µ:

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Menu Value

Codec: G711 µ-law

Payload Size: 20 ms


Settings >Media > Media List

Default Media List:

Media Profiles List: G711a

G711µ

Crypto Profile ID: None

Media DSCP: 46

RTCP Mode: RTCP

Dead Call Detection: Disabled

Silence Suppression: Disabled

TLS Profile


Settings >Security > TLS Profiles

Create TLS Profile:

TLS Protocol: TLS 1.2 Only

Mutual Authentication: Enabled

Allow Weak Cipher: Disable

Handshake Inactivity Timeout: 10

The Client Cipher List is automatically updated to display only the ciphers supported for the selected TLS version

Validate Server FQDN: Disabled

Validate Client FQDN: Disabled

Secondary interface


Settings >Node Interfaces > Logical Interfaces > Ethernet 1 IP

Configure Secondary Interface: Disabled

Primary address dedicated for BT/BTIP traffic

From/To SFB <-> Offnet routing BT/BTIP traffic

SIP Server Table

From/To MS Pool –BT/BTIP



Host: MS Pools FQDN/IP address

Port: 5067

Protocol: TLS

TLS Profile: Select the TLS Profile created

above


From/To BT/BTIP-MS Pool



1st Entry: ACME aSBC nominal

Host: ACME aSBC nominal IP address

Port: 5060

Protocol: TCP


2nd Entry: ACME aSBC backup

Host: ACME aSBC backup IP address

Port: 5060 Protocol: TCP



MS Pool to BT/BTIP



Calling Entry:



need



need

Called Entry:

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Menu Value



need



need

BT/BTIP to MS Pool



Calling Entry:



need



need

Called Entry:




format



Call Routing Tables

From MS Pool



MS Pool to BT/TIP entry:

Description: MS Pool to BT/BTIP

Route Priority: 1

Number/Name Transformation Table: MS

Pool to BT/BTIP

Destination Signalling Group: (SIP) From/To

BT/TIP-MS Pool


Media List: Select the Media List created

above

From BT/BTIP



BT/TIP to MS Pool entry:

Description: BT/BTIP to MS Pool

Route Priority: 1

Number/Name Transformation Table:

BT/BTIP to MS Pool

Destination Signalling Group: (SIP) From/To MS Pool-BT/BTIP


Media List: Select the Media List created

above

Signaling Groups

(SIP) From/To MS Pool – BT/BTIP



Description: SIP From/To MS Pool – BT/BTIP

Call Routing Table: From MS Pool

No. of Channels: 60 (Default)

SIP Server Table: From/To MS Pool –BT/BTIP



Listen Ports:5067 /TLS

TLS Profile: Select the TLS Profile created

above

Federated IP/FQDN: MS Pools IP/FQDN

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Menu Value

(SIP) From/To BT/BTIP-MS Pool



Description: SIP Froom/To BT/BTIP-MS Pool

Call Routing Table: From BT/BTIP

No. of Channels: 60 (Default)

SIP Server Table: From/To BT/BTIP –MS Pool




Federated IP/FQDN: ACME aSBC nominal IP

address

ACME aSBC backup IP

address

Configuration Checklist in case of AudioCodes Mediant 800/1000 E-SBC:

Skype for Business Configuration in case of RS-GW (Topology Builder)


Branch Site > SfB Server > Mediation Pools,

right click and Edit properties

Listening ports TLS: 5067 – 5067

Note:

When both VISIT and B2G offer:

Listening ports TLS must be: 5069


Branch Site > SfB Server > Shared components > PSTN gateways, right click and New IP/PSTN Gateway dedicated for BT/BTIP

Then click Next to define root trunk

FQDN of dedicated gateway for BT/BTIP traffic

Specify BT trunk name


SIP Transport protocol: TLS

Associated Mediation Server: Mediation Pool FQDN


Note:

When both VISIT and B2G offer:

Listening ports TLS must be: 5069

Skype for Business Configuration in case of RS-SBA (Topology Builder)


Branch Site > SfB Server > Mediation Pools, right click and Edit properties

Listening ports TCP: 5060 – 5060








Associated Mediation Server: SBA FQDN



Branch Site > SfB Server > Shared components > PSTN gateways, right click and New IP/PSTN Gateway dedicated for E1/analog

PSTN & Analog Trunk:

Branch Site > SfB Server > Shared Components > Trunks, right click and New Trunk

FQDN of dedicated gateway for E1/Analog traffic

Specify PSTN&Analog trunk name



Associated Mediation Server: SBA FQDN


Skype for Business Configuration in case of HQ with GW aboard (Topology Builder)

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Menu Value


Branch Site > SfB Server > Mediation Pools,

right click and Edit properties

Listening ports TCP: 5060 – 5060








Associated Mediation Server: MS Pool FQDN


AudioCodes Mediant 800/1000 E-SBC configuration

TLS Context

On the AudioCodes Mediant WebUi Interface:

(Advanced mode)

System > TLS Context

Links Tab

TLS Context Certificate

TLS Context Trusted Certificates

Media

Voice Settings


(Advanced mode)

Configuration >VoIP > Media > Voice Settings

Silence Suppression: Disable

DTMF Transport Type: RFC 2833 Relay DTMF

Media Security


(Advanced mode)

Configuration >VoIP > Media > Media Security

Media security: Enable

RTP / RTCP Settings


(Advanced mode)

Configuration >VoIP > Media > RTP / RTCP Settings

RTP Base UDP Port: 16400

Application Enabling

Application Enabling


(Advanced mode)

Configuration >VoIP > Application Enabling > Application Enabling

SBC Application: Enable

Coders and Profiles

Coders


(Advanced mode)

Configuration >VoIP > Coders and Profiles > Coders

Coders Table

Coder Name : G711A-law

Packetization time : 20

Rate : 64

Payloed Type : 8

Silence Suppression : Disabled

Coder Name : G711U-law


Rate : 64

Payload Type : 0


Coders Group Settings

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Menu Value


(Advanced mode)

Configuration >VoIP > Coders and Profiles > Coders Group Settings

Coders Group ID

Coder Name : G711A-law


Rate : 64

Payloed Type : 8


Coder Name : G711U-law


Rate : 64

Payload Type : 0


IP Profile Settings


(Advanced mode)

Configuration >VoIP > Coders and Profiles > IP Profile Settings

SBA or SfB IP Profile ID

(GW tab)

Early Media : Enable

Hold : Enable

(SBC Media tab)

Extension Coders : Coders Group

Allowed Audio Coders : Coders Group

Allowed Coders Mode : Restriction and Preference

BTIP IP Profile ID

(GW tab)

Early Media : Enable

Hold : Enable

(SBC Media tab)

Extension Coders : Coders Group

Allowed Audio Coders : Coders Group

Allowed Coders Mode : Restriction and Preference

VoIP Network

Media Realm Table


(Advanced mode)

Configuration > VoIP > VoIP Network > Media Realm Table

Skype Media Realm (SBA or SfB)

Name : MRm for Skype

IPv4 Interface Name : Mediant IPv4 Interface

Port Range Start : 16900

Number of Media Session Legs : 50

Port Range End : Filled automatically

Default Media Realm : Yes

BTIP Media Realm

Name : MRm for BTIP





Default Media Realm : No

This range is used to accept incoming traffic from

SBC in case of BTIP incoming calls, the defined

range respects the OBS infra recommandations

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Menu Value

SRD Table


(Advanced mode)

Configuration > VoIP > VoIP Network > SRD Table

Name : DefaultSRD

SIP Interface Table


(Advanced mode)

Configuration > VoIP > VoIP Network > SIP Interface Table

One SIP Interface Table for RS SBA

Name : SIPInterface_BTIP&SBA

SRD : DefaultSRD

Network Interface : Mediant IPv4 Interface

Application Type : SBC

TCP Port : 5060

One SIP Interface Table for HQ with GW aboard

Name : SIPInterface_BTIP&SBA

SRD : DefaultSRD



TCP Port : 5060

Two SIPs Interfaces Tables for RS GW

Name : SIPInterface_SfB

SRD : DefaultSRD



TLS Port : 5067

TLS Context Name : TLS Context

Name : SIPInterface_BTIP

SRD : DefaultSRD



TCP Port : 5060

Proxy Set Table


(Advanced mode)

Configuration > VoIP > VoIP Network > Proxy Set Table

Proxy Set Table for Skype traffic (SBA or SfB)

Name : ProxySet for Skype Traffic

SRD : DefaultSRD


SBC IPv4 SIP Interface : SIP Interface for Skype Traffic

Proxy Load Balancing Method : Round Robin

Proxy Keep-Alive Time : 60

Proxy Keep-Alive : Using OPTIONS

(Proxy Address Table)

1 Entries : FQDN or @IP of SBA:5060 TCP (for SBA)

X Entries : FQDN or @IPs of Mediation Pool:5060 TCP (for HQ with GW aboard)

X Entries : FQDN or @IPs of Mediation Pool:5067 TLS (for SfB)

Proxy Set Table for BTIP Traffic

Name : ProxySet for BTIP Traffic

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Menu Value

SRD : DefaultSRD


SBC IPv4 SIP Interface : SIP Interface for BTIP Traffic





2 Entries : FQDN or @IP of aSBC ACME:5060 TCP

IP Group Table


(Advanced mode)

Configuration > VoIP > VoIP Network > IP Group Table

IP Group Table for Skype traffic (SBA or SfB)

Name : IP Profile for Skype Traffic

Type : Server

Proxy Set : Proxy Set for Skype Traffic

IP Profile : IP Profile for Skype Traffic

Media Realm : Media Realm for Skype traffic

IP Group Table for BTIP traffic

Name : IP Profile for BTIP Traffic

Type : Server

Proxy Set : Proxy Set for BTIP Traffic

IP Profile : IP Profile for BTIP Traffic

Media Realm : Media Realm for BTIP traffic

SIP Definitions

General Parameters


(Advanced mode)

Configuration > VoIP > SIP Definitions > General Parameters

PRACK Mode : Supported

Channel Select Mode : Cyclic Ascending

Enable Early Media : Enable

SBC

Allowed Audio Coders Group


(Advanced mode)

Configuration > VoIP > SBC > Allowed Audio Coders Group

Allowed Audio Coders Group ID

Coder Name 1 : G711A-Law

Coder Name 2 : G711U-Law

IP-to-IP Routing Table


(Advanced mode)

Configuration > VoIP > SBC > IP-to-IP Routing Table

SIP Options rule

Name : SIP Options

Alternative Route Options: Route Row

Source IP Group : Any

Request Type : OPTIONS

Destination Type : Dest Address

Destination IP Group : None

Destination SIP Interface : None

Destination Address : internal

Skype to BTIP rule

Name : Skype to BTIP


Source IP Group : Skype IP Group

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Menu Value

Request Type : All

Destination Type : IP Group

Destination IP Group : BTIP IP Group

Destination SIP Interface : BTIP SIP Interface

BTIP to Skype rule

Name : BTIP to Skype


Source IP Group : BTIP IP Group

Request Type : All

Destination Type : IP Group

Destination IP Group : BTIP IP Group

Destination SIP Interface : Skype SIP Interface

Gateway for PSTN calls (Annex 1) Only for RS SBA and RS GW

Trunk Group


(Advanced mode)

Configuration > VoIP > Gateway > Trunk Group

Configure Group Index

Module : PRI

From/To Trunk : 1

Channels : 1-31

Phone Number : Phone number used for the Trunk

Trunk Group ID : Trunk Group ID associated

Trunk Group Settings


(Advanced mode)

Configuration > VoIP > Gateway > Trunk Group Settings

Add Trunk Group Settings

Name : E1 PSTN

Trunk Group ID : Trunk Group ID associated

Channel Selected Mode : Cyclic Descending

Registration Mode : Don’t Register

Trunk Settings


(Advanced mode)

Configuration > VoIP > PSTN > Trunk Settings

Protocol Type : E1 EURO ISDN

Line Code : HDB3

Framing Method : Extend super Frame

VoIP Network Configuration

Media Realm Table


(Advanced mode)


Can be the same as Skype Media Realm







SRD Table


(Advanced mode)


Same as Skype SRD Table

Name : DefaultSRD

SIP Interface Table


(Advanced mode)

Configuration > VoIP > VoIP Network > SIP

SIP Interface Table

Name : SIPInterface_PSTN

SRD : DefaultSRD

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Menu Value

Interface Table Network Interface : Mediant IPv4 Interface for E1/Analog

Application Type : GW

TCP Port : 5060

Proxy Set Table


(Advanced mode)


Proxy Set Table for PSTN traffic

Name : ProxySet for PSTN Traffic

SRD : DefaultSRD

Network Interface : Mediant IPv4 Interface for E1/Analog

SBC IPv4 SIP Interface : SIP Interface for PSTN Traffic





1 Entry : FQDN or @IP of SBA:5060 TCP

IP Group Table


(Advanced mode)


IP Group Table for Skype traffic

Name : IP Profile for PSTN Traffic

Type : Server

Proxy Set : Proxy Set for PSTN Traffic


Media Realm : Media Realm for Skype Traffic

Routing

General Parameters


(Advanced mode)

Configuration > VoIP > Gateway > Routing > General Parameters

Enable Alt Routing Tel to IP : Enable

IP To Trunk Group Routing


(Advanced mode)

Configuration > VoIP > Gateway > Routing > IP To Trunk Group Routing

Skype To PSTN rule

Name : Skype To PSTN


Source SIP Interface : PSTN SIP Interface

Trunk Group ID : PSTN Trunk Group ID

Destination Type : Trunk Group

TEL To IP


(Advanced mode)

Configuration > VoIP > Gateway > Routing > TEL To IP

PSTN To Skype rule

Name : PSTN To Skype

Source Trunk Group ID : PSTN Trunk Group ID

Destination IP Group : Skype IP Group

SIP Interface : PSTN SIP Interface

IP Profile : Skype IP Profile

Gateway for Analog calls (Annex 2)

Trunk Group


(Advanced mode)

Configuration > VoIP > Gateway > Trunk Group

Configure Group Index

Module : FXS

Channels : 1

Phone Number : Analog number in e164 format

Trunk Group ID : Trunk Group ID for Analog

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Menu Value

Trunk Group Settings


(Advanced mode)

Configuration > VoIP > Gateway > Trunk Group Settings

Add Trunk Group Settings

Name : Analog

Trunk Group ID : Trunk Group ID for Analog

Channel Selected Mode : By Dest Phone Number

Registration Mode : Don’t Register

Analog Settings


(Advanced mode)

Configuration > VoIP > Media > Analog Settings

Analog Metering Type : 12 Khz Sinusoidal bursts

FXS Coefficient Type : Europe

VoIP Network Configuration

Media Realm Table


(Advanced mode)


Can be the same as Skype Media Realm







SRD Table


(Advanced mode)


Same as Skype SRD Table

Name : DefaultSRD

SIP Interface Table


(Advanced mode)

Configuration > VoIP > VoIP Network > SIP Interface Table

SIP Interface Table

Name : SIPInterface_Analog

SRD : DefaultSRD


Application Type : GW

TCP Port : 5060

Proxy Set Table


(Advanced mode)


Proxy Set Table for Analog traffic

Name : ProxySet for Analog Traffic

SRD : DefaultSRD


SBC IPv4 SIP Interface : SIP Interface for Analog Traffic





1 Entries : FQDN or @IP of SBA:5060 TCP

IP Group Table


(Advanced mode)


IP Group Table for Skype traffic

Name : IP Profile for Analog Traffic

Type : Server

Proxy Set : Proxy Set for Analog Traffic


Media Realm : Media Realm for Skype Traffic

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Menu Value

Manipulations



(Advanced mode)

Configuration > VoIP > Gateway > Manipulations > IP To Trunk Group Routing

Skype To Analog manipulation rule

Name : Skype To Analog


Destination Prefix : Analog phone number

TEL To IP


(Advanced mode)

Configuration > VoIP > Gateway > Manipulations > TEL To IP

Analog To Any manipulation rule

Name : Analog To Any

Source Trunk Group ID : Analog Trunk Group ID

Destination IP Group : Any

Prefix to Add : +

Routing



(Advanced mode)

Configuration > VoIP > Gateway > Routing > IP To Trunk Group Routing

Skype To Analog routing rule

Name : Skype To Analog


Source SIP Interface : Analog SIP Interface

Destination Phone Prefix : Analog number in e164

Destination Trunk Group : Trunk Group

Trunk Group ID : 2

TEL To IP


(Advanced mode)

Configuration > VoIP > Gateway > Routing > TEL To IP

Analog To Skype routing rule

Name : Analog To Skype

Source Trunk Group ID : Analog Trunk Group ID

Destination IP Group : Skype IP Group

SIP Interface : Analog SIP Interface

IP Profile : Skype IP Profile

CAC Configuration Checklist

CAC Configuration

Enable CAC

SFB PowerShell


Set-CsNetworkConfiguration -EnableBandwidthPolicyCheck

SFB Control Panel

On the Skype for Business control panel interface:

Network Configuration >Global

SFB PowerShell

EnableBandwidthPolicyCheck parameter has to be set to 1

SFB Control Panel

Enable call admission control parameter has to be checked

Media bypass configuration (In case of RS SBA and/or RS Default)

SFB PowerShell

SFB PowerShell

AlwaysByPass parameter has to be

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Menu Value


$a= New-CsNetworkMediaBypassConfiguration -

alwaysByPass $false -Enabled $false

Set-CsNetworkConfiguration –MediaBypassSettings $a

SFB Control Panel



set to false

Enable parameter has to be set to false

SFB Control Panel

Enable media bypass parameter must not be checked

Media bypass configuration (In case of RS GW or a mix of RS GW, RS SBA and RS Default)

SFB PowerShell


$a= New-CsNetworkMediaBypassConfiguration -

alwaysByPass $ false -Enabled $true

Set-CsNetworkConfiguration –MediaBypassSettings $a

SFB Control Panel



SFB PowerShell

AlwaysByPass parameter has to be set to false

Enable parameter has to be set to true

SFB Control Panel

Enable media bypass parameter has to be checked

Choose “Use sites and region

configuration”

Media bypass Trunk Configuration (Only in case of RS-GW)

SFB Control Panel

On the Skype for Business Control panel interface


And then select the RS-GW Trunk to edit Trunk configuration

SFB Control Panel

Enable media bypass parameter has to be checked






Network Region

SFB PowerShell


New-CsNetworkRegion –Identity <XdsIdentity> -CentralSite <Central_Site> –AudioAlternatePath $False -Description “All Locations”

SFB Control Panel



SFB PowerShell

-Identity: The name of the network region

-Central site: The name of the central site

as defined on SFB topology builder

SFB Control Panel

Identity: The name of the network region

Central site: The name of the central site as

defined on SFB topology builder

Audio alternate path: Recommended to disable

Bandwidth Policy profiles

CAC Onnet – Network sites and Network Region CAC

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Menu Value

SFB PowerShell


New-CsNetworkBandwidthPolicyProfile -Identity <BWname> –

Description “Descr Name” -AudioBWLimit

<AudiototalBW> -AudioBWSessionLimit

<AudiosessionBW> -VideoBWLimit <VideototalBW> -

VideoBWSessionLimit <VideoSessionBW>

SFB Control Panel


Network Configuration >Bandwidth Policy

SFB PowerShell

-Identity: The name of the bandwidth region (eg: CAC_basse)

-AudioBWLimit: The total bandwidth

allowed for calls on network sites associated to this BW profile policy

-AudioBWSession Limit: The session

bandwidth allowed for one call on network site associated to this BW profile policy has to be set to 100

-VideoBWLimit: Not applied with BT/BTIP

(used for onnet calls refer to B2G documentation)

-VideoBWSessionLimit: Not applied with

BT/BTIP (used for onnet calls refer to B2G documentation)

SFB Control Panel

Identity: The name of the bandwidth region (eg: CAC_basse)

AudioBWLimit: The total bandwidth


AudioBWSession Limit: The session


VideoBWLimit: Not applied with BT/BTIP


VideoBWSessionLimit: Not applied with


on SFB topology builder

CAC SIP Trunk – Inter site CAC

SFB PowerShell







SFB Control Panel



SFB PowerShell

-Identity: The name of the bandwidth region (eg: CAC_SIPTrunk)









SFB Control Panel

Identity: The name of the bandwidth region (eg: CAC_SIPTrunk)


allowed for BT/BTIP calls on network sites associated to this BW profile policy


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Menu Value

bandwidth allowed for one BT/BTIP call on network site associated to this BW profile policy has to be set to 97






CAC Zero – BT/BTIP network site to Network region CAC

SFB PowerShell







SFB Control Panel



SFB PowerShell

-Identity: The name of the bandwidth region (eg: CAC_Zero)


allowed for calls on network sites associated to this BW profile policy parameter has to be set to 0







SFB Control Panel

Identity: The name of the bandwidth region (eg: CAC_Zero)


allowed for BT/BTIP calls on network sites associated to this BW profile policy parameter has to be set to 0








CAC Edge – Edge network site to Network region CAC

SFB PowerShell







SFB PowerShell

-Identity: The name of the bandwidth region (eg: CAC_Edge)


allowed for calls on network sites associated to this BW profile policy parameter has to be set to 9999999999



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Menu Value

SFB Control Panel







SFB Control Panel

Identity: The name of the bandwidth region (eg: CAC_Edge)


allowed for BT/BTIP calls on network sites associated to this BW profile policy parameter has to be set to 999999999








Network Sites

SFB PowerShell


New-CsNetworkSite-NetworkSIteID <NSname> –Description

“Descr Name” -NetworkRegionID <NRname> -

BWPolicyProfileID <BWPname>

SFB Control Panel


Network Configuration > Site

SFB PowerShell

-NetworkSiteID: The name of the network

site

-Description: Optional

-NetworkRegionID: Select the network region to associate to created network site

-BWPolicyProfileID: Select the bandwidth

profile policy to associate to created network site

SFB Control Panel

-NetworkSiteID: The name of the network site

-Description: Optional

-NetworkRegionID: Select the network region to associate to created network site


profile policy to associate to created network site

Inter Site Policy

SFB PowerShell


New-CsNetworkInterSitePolicy–Identity

<NetworkInterSitename>-BWPolicyProfileID

<SIPTRUNK_BWPname> -NetworkSiteID1 <NS1name>-

NetworkSiteID2 <BTIP_NS_name>

SFB PowerShell

-Identity: The name of the network inter site policy


profile policy to associate to created network inter site policy

-NetworkSiteID1: parameter has to

correspond to the network site 1 (SFB component) to associate to BTIP using inter site policy

-NetworkSiteID2: parameter has to

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Menu Value

correspond to the BT/BTIP network site name

WARNING: NO Inter site for Remote site Gateway

Subnets

SFB PowerShell


New-CsNetworkSubnet-SubnetID <firstsubnetIPaddress>-

MaskBits <maskwo/> -NetworkSiteID <associated

NS_name>

SFB Control Panel


Network Configuration > Subnet

SFB PowerShell

-SubnetID: The first IP address of the corresponding subnet

-MaskBits: The subnet mask to associate to subnet to create without / (eg:32)

-NetworkSiteID: Select the network site

name from the drop down list to associate to this subnet (eg: BTIP)

SFB Control Panel

-SubnetID: The first IP address of the corresponding subnet

-MaskBits: The subnet mask to associate to subnet to create without / (eg:32)

-NetworkSiteID: Select the network site

name from the drop down list to associate to this subnet (eg: BTIP)

Configuration requirements (warnings)

Configuring Clients ports range for LPE and SoftPhone

SFB PowerShell

On the Skype for Business PowerShell Interface

Set-CsConferencingConfiguration –ClientMediaPortRangeEnabled

$true –ClientAudioPort 50060 –ClientAudioPortRange 48

SFB PowerShell

-ClientMediaPortRangeEnable : must be enabled in order to use the specific range

-ClientAudioPort : corresponds to the first

port used for audio

-ClientAudioPortRange : corresponds to the audio range

Configuring Clients ports range for VVX

Using VVX Web UI :

- Navigate through the VVX Web Interface: http:<VVX_IP_Address>

- Go to Settings tab > Network menu > RTP

- Configure the Port Range Start to: 50060

VVX WebUI

Using VVX configuration file (.cfg)

- Configure the following line in the VVX configuration file :

tcpIpApp.port.rtp.mediaPortRangeStart="50060"

- Import the new configuration file to the VVX using the WebUI or through the IIS server

VVX WebUI

or

IIS Server

Others Devices

Check that the audio range port respect the OBS recommendations

The default audio range is: 50060-50107.

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Date post:	27-Aug-2018
Category:	Documents
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